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IETF Session Initiation Protocol

The Session Initiation Protocol (SIP) is a lightweight signaling protocol developed by Henning Schulzrinne at Columbia University. It is designed for initiating, modifying, and terminating multimedia sessions such as telephony and video conferences. SIP allows for personal mobility and supports both unicast and multicast conferencing. It operates over UDP or TCP and features methods for call forwarding, transferring, and handling distinctive ringing. SIP's flexibility and text-based format make it a vital component for network communication. For further details, visit http://www.cs.columbia.edu/~hgs/sip.

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IETF Session Initiation Protocol

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  1. IETF Session Initiation Protocol Henning Schulzrinne Columbia University New York, NY

  2. SIP Overview • Lightweight signaling protocol • to be standardized within IETF MMUSIC • invite to new and on-going conferences • multicast and unicast (“telephony”) • personal mobility • call center: reach first, first all • change and add media • post dial delay: 1.5 RTT • uses either UDP or TCP

  3. SIP and H.323

  4. SIP addresses food chain

  5. SIP protocol • text based (~ HTTP) • methods

  6. SIP operation in redirect mode

  7. SIP operation in proxy mode

  8. SIP for Intelligent Networks • IN services “inside” & “outside” network • UPN: translation by redirect server • call forwarding: proxy, end system • call transfer: BYE • distinctive ringing: end system • 800#: charging responsibility of resource reservation • call center (queueing, first available)

  9. For more information... • http://www.cs.columbia.edu/~hgs/sip • IETF MMUSIC mailing list:confctrl-request@isi.edu

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